Add information about microphone gain changes to AEC3

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1127 matching lines @@
   if (log_rms) {
     capture_rms_interval_counter_ = 0;
     RmsLevel::Levels levels = capture_input_rms_.AverageAndPeak();
     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelAverageRms",
                                 levels.average, 1, RmsLevel::kMinLevelDb, 64);
     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelPeakRms",
                                 levels.peak, 1, RmsLevel::kMinLevelDb, 64);
   }
 
   if (private_submodules_->echo_canceller3) {
+    const int new_agc_level = gain_control()->stream_analog_level();
+    capture_.echo_path_gain_change =
+        (capture_.previous_agc_level != new_agc_level);
+    capture_.previous_agc_level = new_agc_level;
     private_submodules_->echo_canceller3->AnalyzeCapture(capture_buffer);
   }
 
   if (constants_.use_experimental_agc &&
       public_submodules_->gain_control->is_enabled()) {
     private_submodules_->agc_manager->AnalyzePreProcess(
         capture_buffer->channels()[0], capture_buffer->num_channels(),
         capture_nonlocked_.capture_processing_format.num_frames());
   }
 
@@ skipping 28 matching lines @@
   public_submodules_->noise_suppression->AnalyzeCaptureAudio(capture_buffer);
 
   // Ensure that the stream delay was set before the call to the
   // AEC ProcessCaptureAudio function.
   if (public_submodules_->echo_cancellation->is_enabled() &&
       !was_stream_delay_set()) {
     return AudioProcessing::kStreamParameterNotSetError;
   }
 
   if (private_submodules_->echo_canceller3) {
-    private_submodules_->echo_canceller3->ProcessCapture(capture_buffer, false);
+    private_submodules_->echo_canceller3->ProcessCapture(
+        capture_buffer, capture_.echo_path_gain_change);
   } else {
     RETURN_ON_ERR(public_submodules_->echo_cancellation->ProcessCaptureAudio(
         capture_buffer, stream_delay_ms()));
   }
 
   if (public_submodules_->echo_control_mobile->is_enabled() &&
       public_submodules_->noise_suppression->is_enabled()) {
     capture_buffer->CopyLowPassToReference();
   }
   public_submodules_->noise_suppression->ProcessCaptureAudio(capture_buffer);
@@ skipping 779 matching lines @@
       was_stream_delay_set(false),
       last_stream_delay_ms(0),
       last_aec_system_delay_ms(0),
       stream_delay_jumps(-1),
       output_will_be_muted(false),
       key_pressed(false),
       transient_suppressor_enabled(transient_suppressor_enabled),
       array_geometry(array_geometry),
       target_direction(target_direction),
       capture_processing_format(kSampleRate16kHz),
-      split_rate(kSampleRate16kHz) {}
+      split_rate(kSampleRate16kHz),
+      previous_agc_level(0),
+      echo_path_gain_change(false) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc